1. Field of the Invention
The present invention relates to methods and apparatuses for encoding and decoding, and more particularly, to methods and apparatuses for low bit rate encoding and decoding, which can efficiently compress data at a low bit rate while maintaining high sound quality.
2. Description of Related Art
Information carrier waves are analog signals, which are continuous in time and amplitude. Accordingly, in order to represent the information, carrier waves in a discrete form, analog-to-digital (A/D) conversion is used. A/D conversion comprises two processes: discretion in time (sampling), and quantization of amplitude. Sampling is a process that converts time continuous signals into time discrete signals. Amplitude quantization is a process that defines the number of possible amplitudes of discrete signals. Namely, amplitude quantization replaces input amplitude x(n) by y(n) within a limit of possible amplitude levels.
Generally, digital data is obtained after sampling and amplitude quantization of analog signals. It is then stored in a recording/storage medium, such as a compact disc (CD) or a digital audio tape (DAT), in pulse code modulation (PCM) format to be reproduced as needed. The PCM scheme for storage and reproduction helps to improve sound quality and to prevent degradation over time in comparison with any other analog scheme, but has a problem in the storage and communication of large amounts of data.
To solve this problem of the PCM scheme, differential pulse code modulation (DPCM) and adaptive differential pulse code modulation (ADPCM) schemes have been developed. Using these schemes, attempts have been made to reduce the amount of digital audio data, however, their efficiencies vary greatly depending on signal types. In the Moving Pictures Experts Group (MPEG)/audio scheme, which recently have been standardized by the International Standard Organization (ISO), or in the AC-2/AC-3 scheme, developed by Dolby Laboratories Inc., the human psychoacoustic model has been used to efficiently reduce the amount of data.
In known audio data compression schemes, such as MPEG-1/audio, MPEG-2/audio, or AC-2/AC-3, signals in the time domain, which are grouped into blocks of a set size, are transformed into signals in the frequency domain. The transformed signals are then subjected to scalar quantization using the human psychoacoustic model. The scalar quantization is simple, but not optimal, even when input samples are statistically independent, and it is certain to be at a great insufficiency when input samples are statistically dependent. To compensate for this, lossless compression encoding, such as entropy encoding or another type of adaptive quantization, is incorporated into the encoding process. Consequently, audio data compression schemes become much more complicated than those that only stores PCM data, and have bitstreams containing not only quantized PCM data but also additional information for data compression.
An MPEG/audio standardized scheme or an AC-2/AC-3 scheme provides sound quality comparable to that of a compact disc, at one-eighth to one-sixth of data of other known digital encoding methods, and at a bit rate of between 64 and 384 kbps. Thus, the MPEG/audio standard is expected to play an important role in storing and communicating audio signals in multimedia systems, such as digital audio broadcasting (DAB), audio on demand (AOD), and Internet phones.
Unfortunately, when encoding at low bit rate below 32 kbps, the encoding method with only signal quantization lacks available bits to encode. Accordingly, there is a need to have an efficient method for low bit rate compression of audio signals that can maintain close-to-original sound reproduction.